g722

New problem, it should be video codec there is a problem. Look for it.WEBRTC Voiceengine Codecs:ISAC/16000/1 (103)ISAC/32000/1 (104)Unexpected CODEC:ISAC/48000/1 (105)PCMU/8000/1 (0)PCMA/8000/1 (8)Unexpected CODEC:PCMU/8000/2 (110)Unexpected CODEC:PCMA/8000/2 (118)ILBC/8000/1 (102)G722/8000/1 (9)Unexpected CODEC:G722/8000/2 (119)CN/8000/1 (13)CN/16000/1 (105)CN/32000/1 (106)TELEPHONE-EVENT/8000/1 (126)RED/8000/1 (127)Webrtcvideoengine2::webrtcvideoeng

Webrtc\modules\media_file directory. This feature can be used as a video source with local files, a bit like the function of a virtual camera, and the supported formats are AVI. In addition, WEBRTC can also record audio and video to local files, more useful functions.VideoImage processing--video_processing source code in the Webrtc\modules\video_processing directory. Video image processing for each frame of the image processing, including shading detection, color enhancement, noise reduction pr

coding specifications, and the aac-ld (low delay, lower latency specification) is encoded at low bit rate. It supports 8k~48k sampling rate, can output 64Kbps of the bitrate close to CD quality audio, and supports multi-sound channels, AAC-LD algorithm delay is only 20ms.AAC is more powerful because of its modular design. The frame structure itself can be filled with new things, which makes it possible for the cores of different development to be fused together and absorbed into each other.7. C

supported formats include Avi. In addition, WebRTC can also be used to record audio and video files to local files. Video Image Processing-video_processing The source code is in the WebRTC \ modules \ video_processing directory. Video Image Processing processes each frame of the image, including brightness detection, color enhancement, and noise reduction, to improve the video quality. Video Display -- video_render The source code is in the WebRTC \ modules \ video_render directory.

\video_processing directory.Video image processing for each frame of the image processing, including shading detection, color enhancement, noise reduction processing and other functions to improve video quality.Video Display--video_renderThe source code is in the Webrtc\modules\video_render directory.On the Windows platform, WEBRTC uses Direct3D9 and DirectDraw to display video, only this way, it must.Network Transmission and flow controlFor network video, the transmission and control of data is

delay, lower latency specification) is encoded at low bit rate. It supports 8k~48k sampling rate, can output 64Kbps of the bitrate close to CD quality audio, and supports multi-sound channels, AAC-LD algorithm delay is only 20ms.AAC is more powerful because of its modular design. The frame structure itself can be filled with new things, which makes it possible for the cores of different development to be fused together and absorbed into each other.7. Comparison of the main parameters of various

1. snd_pcm_open: Open the handle. 2. configuration parameters: snd_pcm_hw_params_alloca, snd_pcm_hw_params_any, parameters, parameters, snd_pcm_hw_params_set_access, parameters, parameters, and snd_pcm_hw_params. 3. read/write: snd_pcm_writei and snd_pcm_readi. Note: 1. Create a handle Based on the functions to be implemented. The snd_pcm_open parameter snd_pcm_stream_capture corresponds to snd_pcm_readi, And the snd_pcm_stream_playback corresponds to snd_pcm_writei. 2. Configure parameters

record audio and video to local files, more useful functions.video image processing--video_processingThe source code is in the Webrtc\modules\video_processing directory. Video image processing for each frame of the image processing, including shading detection, color enhancement, noise reduction processing and other functions to improve video quality.Video Display--video_renderThe source code is in the Webrtc\modules\video_render directory. On the Windows platform, WEBRTC uses Direct3D9 and Dir

")) {#ifdef WEBRTC_CODEC_AMRWB return new ACMAMRWB (KGSMAMRWB); #endif} else if (! STR_CASE_CMP (Codec_inst->plname, "G722")) {#ifdef webrtc_codec_g722 return new ACMG722 (kG722); #endif} else if (! STR_CASE_CMP (codec_insT->plname, "G7221")) {switch (codec_inst->plfreq) {case 16000: {#ifdef webrtc_codec_g722_1 int codec_id; Switch (codec_inst->rate) {case 16000: {codec_id = kg722_1_16; Break } case 24000: {co

effect is good. Audio/Video decoding Encoder BV, GSM, speex, PCMU, PCMA, G722H, and silk. H.263 + H263 + H.263 H264-MP H264-BP Theora MP4V-ES VP8 G.729 G.722 Speex-UWB Speex-WB Speex-NBILBC gsm pcmu pcma. H.263, H264, G729, iLBC, speex, and silk ......, The decoder is used as a plug-in. G722, GSM, arm, ilbc, speex, PCMU, PCMA, G722H, silk, vp8, h264, mpeg4 ....... Stun/Turn Technology Only Stun is supported. Suppor

Address: http://www.pjsip.org/pjsua.htm Introduction Pjsua is an open-source command line SIP User proxy (soft phone), which is implemented using pjsip protocol, pjnath, and pjmedia. Although it only has a simple command line interface, it is fully functional. SIP function: Multiple IDs (account registration );Multiple calls;Support IPv6 (added in version 1.2 );Prack (100rel, RFC 3262 );Update (RFC 3311 );Options;Call persistence;Call transfer;Simple pidf and IDF support (subscription/no

the bandwidth is sufficient and the voice quality needs to be better, PCMU and PCMA can be used, and even the broadband encoding method g722 (64 Kbps) can be used, which can provide high-fidelity sound quality. PCMA (g.711a) Type: Audio Maker: ITU-T Required bandwidth: 64 Kbps (90.4) Features: Both PCMU and PCMA provide better speech quality, but they occupy a high bandwidth and require 64 Kbps. Advantage: Excellent speech quality Disadvanta

RFC 3551 9 G722 Audio 1 8000 ITU-T g.722 audio RFC 3551-page 14 10 L16 Audio 2 44100 Linear PCM 16-bit stereo audio 1411.2 kbit/s,[2][3][4]Uncompressed RFC 3551, page 27 11 L16 Audio 1 44100 Linear PCM 16-bit audio 705.6 kbit/s, uncompressed RFC 3551, page 27 12 Qcelp Audio 1 8000 Qualcomm Code Excited Linear Prediction RFC 2658,RFC 3551

payload type. Payload Type 2 is assigned to G721 in RFC 1890 and to it equivalent successor G726-32 in draft versions of this speci CA tion,But it use was now deprecated and that static payload type was marked reserved due to conflicting useFor the payload formats g726-32 and AAL2-G726-32 (see section 4.5.4). Payload type indicatesThe Comfort Noise (CN) payload Format speci ed in RFC 3389 [9]. Payload type is marked\reserved "Because some draft versions of this speci cation assignedVersion of t

) Architecture:Based on Doubango (Doubango is an open-source framework based on 3GPP Ims/rcs and can be used in embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (video), ILBC (audio), pcma,pcm

[Email protected]: ~ Linphonec-v-D 6Info: No logfile, logging to stdoutOrtp-message-oRTP-0.20.0 initialized.Ortp-message-assigning PCMU/8000 payload type number 0Ortp-message-assigning GSM/8000 payload type Number 3Ortp-message-assigning PCMA/8000 payload type number 8Ortp-message-assigning speex/8000 payload type number 110Ortp-message-assigning speex/16000 payload type Number 111Ortp-message-assigning speex/32000 payload type number 112Ortp-message-assigning telephone-event/8000 payload type n

1. audio format: G722 HD voice (64 kbit)-only over WLAN and 3G (requires paid account) PCMA (64 kbit)-only over WLAN and 3G PCMU (64 kbit)-only over WLAN and 3G Speex (11 kbit)-always try GSM (13 kbit)-always try Bv16 (16 kbit)-always try From the source code, g711 exists, but g729. 2. Gain: headset, microphone, headset 3. video quality: High 352x288 @ 360 kbit Low 176x144 @ 192 kbit Sipdroid currently supports basic call functions. Special PBX functi

do not test.II) imsdroid1) Architecture:Based on Doubango (Doubango is an open-source framework based on 3GPP Ims/rcs and can be used in embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (vide

Reprint Annotated Source Http://blog.csdn.net/wanghorse├──./Base //Base Platform library, including threads, locks, sockets, etc.├──./build//Compile script, Gyp├──./common_audio//Basic public audio processing│├──./common_audio/include//On a type conversion header file│├──./common_audio/resampler//Audio resampling code│├──./common_audio/signal_processing//Audio signal processing code, related to hardware platform, with assembly code│└──./common_audio/vad//vad Code├──./common_video//Basic public

Reprint Annotated Source Http://blog.csdn.net/wanghorse├──./Base //Base Platform library, including threads, locks, sockets, etc.├──./build//Compile the script. Gyp├──./common_audio//Basic public audio processing│├──./common_audio/include//On a type conversion header file│├──./common_audio/resampler//Audio re-sample code│├──./common_audio/signal_processing//Audio signal processing code, related to hardware platform, with assembly code│└──./common_audio/vad//vad Code├──./common_video//Basic publ